1. Field of the Invention
This invention relates generally to transmission and switching techniques in telephone communication systems and, more particularly, to an improved conference technique whereby a number of channels in a telephone switching system employing pulse code modulation for transmission purposes are combined so that a number of subscribers may participate in a common telephone conversation. More particularly still, it relates to improvements in a multi-port conference circuit of the type disclosed in U.S. patent application Ser. No. 857,168 filed Dec. 5, 1977, now U.S. Pat. No. 4,175,215 issued Nov. 20, 1979, which is assigned to the same assignee as the present invention.
The present invention pertains to a multi-port conference circuit for use in a private automatic branch exchange similar to those units manufactured by GTE Automatic Electric Incorporated and designated GTD120. Circuitry with minimum modification could also be employed in class five central offices that employ digital switching. Such telephone systems employ a time switching network rather than a space divided switching network.
In time division switching networks a requirement exists to have sources of pulse code modulated voice samples associated with time slots. These time slots allow the conference to sequentially receive the code for each conferee. For the conference circuit to be effective, it must be able to recognize who the conferees are and, of course, who is not associated with the conference. The circuitry must also be capable of distributing the conference speakers' code to each conferee. Information of this sort is, of course, available in the telephone switching systems referred to above. It should be understood that only telephone switching systems employing pulse code modulation can use the circuitry of the present invention, and such circuitry interfaces with time division portions of such switching networks.
2. Description of the Prior Art
An approach to the handling of pulse code modulated information and conference circuitry is taught by U.S. Pat. Nos. 3,699,264, 4,007,338 and 4,054,755, which are assigned to the same assignee as the present invention. In these noted patents, digital signals are not converted to analog; but rather the binary words from the participating channels are compared with the channel having the smallest binary numbers selected as the speaker. An improvement in the conference circuitry disclosed in these above-identified U.S. patents is disclosed in the above-referenced U.S. patent application Ser. No. 857,168 now U.S. Pat. No. 4,175,215.
PCM conferencing as taught in the above-identified patents and application requires a source of pulse code modulated (PCM) coded voice samples which have associated time slots. These time slots allow the conference to sequentially receive a code for each conferee. The conference circuitry must be able to recognize who the conferees are and who is not associated with the conference call.
In the above-referenced U.S. patent application Ser. No. 857,168 now U.S. Pat. No. 4,175,215 PCM samples are taken for each conferee from the time switch and via comparator circuits, a PCM sample is sent to the conferee. Since the selected PCM sample is not determined until all samples are compared, a frame delay is required after which all conferees except the selected conferee will receive the selected PCM sample from the previous frame. The selected conferee, in turn, receives a null code (perfect idle channel). To minimize speech clipping or selecting noise, two circuits, a preliminary and a preferred speaker preference circuit, are employed.
The preliminary preference circuit utilizes the identity of the previous selected speaker and after its PCM sample is compared, its binary weight is modified to the highest value of a corresponding curve segment. This is done by adding a bit between the segment and the step bits, allowing the binary value to be decreased. This technique permits the conference circuit to hold onto the previous speaker if the incoming PCM samples are in the same PCM segment or below in value.
The preferred speaker preference circuit functions when the magnitude of the present PCM sample exceeds the value of the preferred preference circuit threshold. When a speaker is selected for the succeeding frame and has a larger PAM (smaller PCM code) sample than the threshold, a preferred preference circuit creates a lower binary weight (apparently larger PAM) to the comparator, for the selected speaker, for a period of one frame. This reduces speech clipping during that time when two or more conferees are conversing simultaneously.
Neither the preliminary nor the preferred preference circuit alters the incoming or the outgoing PCM sample to the comparison circuit to favor the previous speaker.
Further improvements in the multi-port conference circuit taught in the above-referenced U.S. patent application Ser. No. 857,168 now U.S. Pat. No. 4,175,215 are directed to reduce or substantially eliminate the problem of high idle channel noise resulting from always choosing the largest signal above null code (quiet or absence of signal), the distortion of signals to the listeners and distortion of the speaker side tone, and finally difficulties from foreign signals.
A major contribution to the speech quality degradation in digital conferencing is due to signal reflections of the original signal circulating and fighting for control of the conference. To provide transmission of only the primary signal, a continuous threshold is established to pass the primary and exclude the reflection. It is only used in the selection process. For conditions which do not provide the threshold being met, the previous speaker is retained.
Multiple speaker operation still provides flip-flop operation and is very rapid as compared to echo suppressor type flip-flopping or speaker phone operation. Thus, the loss of syllables is not heard, however, one may notice the shift of background noise levels, especially if one has a background signal like a radio.
This constant switch is reduced, by a locking method where though the foreign signal is still present, it is not chopped up due to switching between idle channels.
The conference circuitry includes two threshold comparators, one for the new conferee and one for the temporary speaker.
The new conferee's PCM is measured against the threshold, as well as that which reaches a temporary speaker PCM buffer which is also sensed to see if it is the previous speaker. Only conferees with PCM codes greater than that of the temporary speaker are allowed to take over. A greater code is actually of less binary value so A&lt;B allows the update to exist. The A corresponds to the conferee PCM buffer when the B corresponds to the temporary speaker PCM buffer. If the conferee is the previous speaker, it is updated unless the temporary speaker signal is greater and also exceeds the threshold. Once in the temporary speaker PCM buffer, only conferees which meet the threshold and exceed the temporary speaker PCM buffer will be allowed to take over. If neither of the conditions occur (i.e., the previous speaker is not encountered for this decision) just the PCM code values are used with the last being allowed to take over the temporary buffer.